search for: snom300

hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ====ok inbound and outbound the calls between x-lite and snom3...

...ending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and ulaw) - PSTN Peer (supports g729 and ulaw) - Remote Asterisk Peer (supports speex and ulaw) Currently, it's configured like this: [snom300] disallow=all allow=ulaw [pstnpeer] disallow=all allow=ulaw [asteriskpeer] disallow=all allow=speex which translates to this: Snom300 ---ulaw---> (pass-thru) ---ulaw----> PSTNPeer Snom300 ---ulaw---> (transcode) ---speex---> AsteriskPeer In other words, my S...

...;-- SIP read from 172.28.20.4:2051: INVITE sip:374@172.28.2.30;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone> Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/6.2.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: tal...

please no body has som with aastra can help me in this issue 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit <salah.elharit200 at gmail.com>: > hello list > > i need your help please regarding an issue with snom300 and aastra6731i > using asterisk > > 11.13.0 asterisk > > snom 300 8.7.3.25 > > astra 6731i 2.6.0.2019 > > i have configured the trunks like below > > 100 in snom300 > 200 in snom300 > 300 in aastra6731i > 400 in x-lite > > the calls between x-lite...

...ll-ID: 3c26701f2ede-afeuhg58c60m CSeq: 7 REGISTER Max-Forwards: 70 Contact: <sip:361 at 192.168.101.102:2061;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom300/8.4.31 Allow-Events: dialog X-Real-IP: 192.168.101.102 Supported: path, gruu Expires: 3600 Content-Length: 0 <-------------...

...> please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit > <salah.elharit200 at gmail.com <mailto:salah.elharit200 at gmail.com>>: > > hello list > > i need your help please regarding an issue with snom300 and > aastra6731i using asterisk > > 11.13.0 asterisk > > snom 300 8.7.3.25 > > astra 6731i 2.6.0.2019 > > i have configured the trunks like below > > 100 in snom300 > 200 in snom300 > 300 in aastra6731i > 400 in x-lite...

...or so now. Recently it lost all its settings and I had to reconfigure it via the built in website. For a few weeks it was fine. Couple of days ago it lost its settings again. I logged in to its web server and thought I would upgrade the firmware. It seems to be running an old version: Phone Type: snom300-SIP MAC-Address: 00041325244C IP-Address: 192.168.1.254 Application-Version: snom300-SIP 6.0.3 Rootfs-Version: snom300 jffs2 v3.36 Firmware-URL: Production Information: Mac:00041325244C;Version:Standard;Hardware:snom300 (MB V3.2_A11);Date: 04.04.06;Copyright(C) snom technology AG I put in the UR...

...th Asterisks for few years. They used go occasionaly NR, and I would reregister them from the phone web interface. But it started doing frequently for last few months. Here are SNOM Phone and the firmware version; snom190-SIP - Version-Code: snom190-SIP 3.56m snom320-SIP - snom320 jffs2 v3.36 snom300-SIP - snom300-SIP 6.5.2 Asterisk version - Asterisk CVS-01/29/04-16:41:27 I would appreciate any help to fix the NR problem. Thanks

Hi all, I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk through OpenVPN seems to have the problem. From CDR, I see for 3 calls from this morning I'm aware of, that asterisk hangup after respectively 899s 894s 898s In logs I see WARNING[8213] chan_sip.c: Retransmission timeout reached on transmission 522eec628683-uy8...

...17:05, Salaheddine Elharit wrote: > > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit <salah.elharit200 at gmail.com > >: > >> hello list >> >> i need your help please regarding an issue with snom300 and aastra6731i >> using asterisk >> >> 11.13.0 asterisk >> >> snom 300 8.7.3.25 >> >> astra 6731i 2.6.0.2019 >> >> i have configured the trunks like below >> >> 100 in snom300 >> 200 in snom300 >> 300 in aastra673...

Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun -------------- next p...

Hi list, We have a strange behavior: a customer Snom300 behind a public FW has contact like contact              : sip:user at x.y.39.147:2048;x-ast-orig-host=169.254.252.1:2048 The phone can place calls but not receive any. Also, qualify give unreachable which seems correct when looking the x-ast-orig-host IP. Problem is that the local IP of this...

...al extensions, a zap trunk and a working sip trunk with sipgate.de. My asterisk server is behind a NAT/Firewall, anyhow it registers and works well with sipgate.de on incoming and outgoing calls. I aquired an account with a reseller net-voz.com: I did some testing with the account directly from a Snom300 phone - works without a problem, (behind the nat) I spent hours testing and adjusting the trunk configuration for net-voz, maybe sombody on the list can give a helpful hint: First of all: Registry works! pbx*CLI> sip show registry Host Username Refresh State R...

...Content-Type: application/pidf+xml Content-Length: 480 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:im="urn:ietf:params:xml:ns:pidf:im" entity="pres:1001 at example.com"> <tuple id="snom300-0004133D6914"> <status> <basic>open</basic> <im:im>Available</im:im> </status> <contact priority="1.00">sip:1001 at example.com</contact> <note xml:lang="en">Available</note> </tuple> </presence> ##...

Hi, I've configured my exten.conf for few exten. But I'm curious to know how long can be my exten like (exten => XXXXXXX.....). Is there any limit for this or not. B'coz I've noticed one strange problem. I'm usnig snom300 as my hard phone to make calls. when my exten length is 14 then calls goes immed. without any problem but when I change length from 14 to 15 call goes but when I dial 10 times I get only 1 or 2 connect (that is call never lends on my server if length is 15) but if I change length to 14 then 10/10 c...

...'m getting this error in the logs: "That's odd... Got a response on a call we dont know about" I see it's part of the SIP Channel debugging, something is wrong with the response headers to a request. I get the message every time it opens a channel to a Snom Technologies ag Snom300 phone. Any help much appreciated! emdeex Jun 5 13:18:22 DEBUG[26785] acl.c: ##### Testing 124.176.197.132 with 192.168.0.0 Jun 5 13:18:22 DEBUG[26785] chan_sip.c: Target address 124.176.197.132 is not local, substituting externip Jun 5 13:18:22 DEBUG[26785] chan_sip.c: Allocating new SIP dialo...

...asterisk group features to manage group and category of a sip channel. I have done a lot of test about it but always it doesn't work correctly if I transfer the call. This is the macro code I use for inbound calls. [macro-test] ; ${ARG1} - technology something like SIP ; ${ARG2} - resource. snom300-for-vasya ; ${ARG3} - dial timeout ; ${ARG4} - dial options ; ${ARG5} - dial url exten => s,1,Goto(s-set-variables,1) exten => s,n(set_var_ret),Set(GROUP(${LOCAL_PARTY})=OUTBOUND_GROUP) exten => s,n,GotoIf($[${GROUP_COUNT(OUTBOUND_GROUP@${LOCAL_PARTY})} > 1]?play_back_busy) exten =&gt...

...: (none) Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw:20,alaw:20) Auto-Framing : No Status : OK (23 ms) Useragent : snom320/8.7.3.19 Reg. Contact : sip:blabla0 at 10.0.12.21:2067 As you see, Def. Username should be tel-221 and not Reception1 ! We have around 20 SNOM300 and 320, only with this one we have this problem. Firmware version is the same in other SNOM320. We restart Asterisk, the server himself, the phone, no changes. Reception1 is the defaultuser we used previously and which is no more existing in any configuration file. Thanks for any hint. -- D...

...MY_PHONE_NUMBER>@sipgate.de;user=phone> > Call-ID: 4fdf703d880d-ywqwnfbbj1h7 > CSeq: 2 INVITE > Max-Forwards: 67 > Contact: > <sip:<CALLING_PARTY_SIP_ID>@<CALLING_PARTY_IP_ADDRESS>:2048;line=swnt2d3t>;reg-id=1 > X-Serialnumber: 000413251D76 > User-Agent: snom300/8.7.3.7 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, > MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, 100rel, replaces, from-change > Session-Expires: 3600;refresher=uas > Min-SE:...